search for: gincantalupo

...l in order to check sip messages. I have to sniff both asterisk and zoiper sip messages. I know that this can be tricky but this can help you to understand what is wrong in sip messages. Please let me know if you need more detail. Marino On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo < gincantalupo at fgasoftware.com> wrote: > Hi Marino, > > I tried to connect zoiper directly to the provider with the same account > parameters I'm using with Asterisk. Zoiper connects without problems. It > is true tnet.it is not resolvable but I can use the proxy URL > sip.tnet.it whi...

...if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at fgasoftware.com> wrote: > Hi Marino, > Asterisk gives a timeout on registration and a "no such host" because > cannot resolve tnet.it but that server address is not resolvable so I > think that is not a problem (my zoiper connects to the provider without > problems,...

Hi All, I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX 2.7.0.10. I have configured x100p fxo card in my asterisk box. But in my cli mode i am not getting the command *"dahdi show status"* Output of CLI : astrisks*CLI> *dahdi show status* No such command 'dahdi show status' (type 'core show help dahdi show' for other possible commands) I

Hi, I'm making some tests with Tellfree brazilian provider. I'm using 2 users A and B, one for calling and the other to receive calls. When I make a call I can see (from the CLI console) user A is calling user B but user B does not answer (the phone continues to ring) even if the "sip show registry" command says user B is registered. In my sip.conf I have: register =>

Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help

Hi, is there anyone trying a power over ethernet solution to feed IP phones? I'd like to buy a "good but cheap" hub/switch but I don't know which. Can anybody help me?? TIA Giorgio

Hi, Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

...Commercial Discussion > Subject: Re: [Asterisk-Users] how to show called name on > calling polycomdisplay > > I was looking for this exactly as well.... > > Any ideas? > > - Gabe > > > ----- Original Message ----- > From: "Giorgio Incantalupo" <gincantalupo@fgasoftware.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Wednesday, March 15, 2006 12:52 AM > Subject: [Asterisk-Users] how to show called name on calling > polycom display > > > >...

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

...#39;/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format* Is there anybody who can help me?? TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

Hi, does anybody know what *zttranscode *module* *is for*?* Thanks!! Giorgio -- _________________________________________________ Giorgio Incantalupo, mailto:gincantalupo@fgasoftware.com FG&A srl - http://www.fgasoftware.com - Voice@Work - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172

Hi, I'm testing attended transfer with 3 SIP phones. I noticed about 10% of my transfers make the call drop and I get this on my log: Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback: Failed to write frame -- Playing 'beep' (language 'it') Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer: Failed to play transfer sound! Moreover, every

Hi all, has anybody ever encountered this ERROR before? It happens frequently on my debian6-based pbx. I'm using Asterisk 1.8.11 with dahdi-linux-2.4.1 and a quadBRI card. ERROR: chan_dahdi.c: PRI Span: 3 PROBLEM: General: Badly Structured Component I tried to google but without success. Do you know what it means? Should I worry? Thank You Giorgio

Hi, I get a lot of these messages on my Asterisk CLI: "Failed to authenticate user 1000<sip:1000 at MY_OWN_IP_ADDRESS>;tag=03f82bb9" as if my PBX machine is trying to authenticate to itself. It seems someone is attacking my asterisk PBX. Is there a way to fix this problem? Thank you. Giorgio Incantalupo

Dear all I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on is asterisk and it is working fine but i got this DTMF error on asterisk CLI what is it ?? -- Zap/36-1 is ringing -- Zap/36-1 answered SIP/5406-9fa59770 -- Channel 0/1, span 2 got hangup request, cause 31 [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to forward voice or

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo

Hi, emails to astricon.net seems to bounce (at least for me) I need information about proper & authorized Asterisk training in the Miami, FL area and the possibility of later DCAP testing. Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

...e two lines of [ins_ext] context after "exten => 103,2,Hangup" extensions 104 can be called) TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid